Cannot route incoming call -- SOLVED

Asterisk Asterisk Support
1

Hello All,

I just setup an elastix server here and for the most part things are going well. :smile:

I have got a couple extensions up and communicating between each other. I have also managed to get asterisk talking to gizmo5 so I can make and receive calls to land lines as well.

I am, unfortunately, running into a problem when I try to route the incoming call. If I point the call directly at an extension, or the voicemail for that extension, the call works perfectly. If I want the incoming call to go to an IVR, ring group, announcement, etc, I get absolute silence. Any key that is pressed results in the call terminating, if I do nothing it just times out. If, however, I use the test call (7777), the system works as intended (IVR w/welcome message).

The incoming call is set to "Any DID / Any CID"
Asterisk version: 1.4.19
Elastix version: 1.0-17

trunk settings in sip_additional.conf:

username=1747xxxxxxx type=friend secret=xxx qualify=yes nat=yes insecure=very host=proxy01.sipphone.com fromuser=1747xxxxxxx fromdomain=proxy01.sipphone.com dtmfmode=rfc2833 disallow=all context=from-pstn canreinvite=no authuser=1747xxxxxxx allow=ulaw allow=alaw allow=ilbc allow=gsm

When I run “asterisk -vvvvvr” and watch the calls come in this is what I get for an incoming call from my cell phone:

    -- Executing [1747xxxxxxx@from-pstn:1] NoOp("SIP/1747xxxxxxx-09bacf30", "Catch-All DID Match - Found 1747xxxxxxx - You probably want a DID for this.") in new stack
    -- Executing [1747xxxxxxx@from-pstn:2] Goto("SIP/1747xxxxxxx-09bacf30", "ext-did|s|1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] Set("SIP/1747xxxxxxx-09bacf30", "__FROM_DID=s") in new stack
    -- Executing [s@ext-did:2] GotoIf("SIP/1747xxxxxxx-09bacf30", "1 ?cidok") in new stack
    -- Goto (ext-did,s,4)
    -- Executing [s@ext-did:4] NoOp("SIP/1747xxxxxxx-09bacf30", "CallerID is "+1xxxxxxxxxx" <+1xxxxxxxxx>") in new stack
    -- Executing [s@ext-did:5] Goto("SIP/1747xxxxxxx-09bacf30", "ivr-2|s|1") in new stack
    -- Goto (ivr-2,s,1)
    -- Executing [s@ivr-2:1] Set("SIP/1747xxxxxxx-09bacf30", "LOOPCOUNT=0") in new stack
    -- Executing [s@ivr-2:2] Set("SIP/1747xxxxxxx-09bacf30", "__DIR-CONTEXT=default") in new stack
    -- Executing [s@ivr-2:3] Set("SIP/1747xxxxxxx-09bacf30", "_IVR_CONTEXT_ivr-2=") in new stack
    -- Executing [s@ivr-2:4] Set("SIP/1747xxxxxxx-09bacf30", "_IVR_CONTEXT=ivr-2") in new stack
    -- Executing [s@ivr-2:5] GotoIf("SIP/1747xxxxxxx-09bacf30", "0?begin") in new stack
    -- Executing [s@ivr-2:6] Answer("SIP/1747xxxxxxx-09bacf30", "") in new stack
    -- Executing [s@ivr-2:7] Wait("SIP/1747xxxxxxx-09bacf30", "1") in new stack
    -- Executing [s@ivr-2:8] Set("SIP/1747xxxxxxx-09bacf30", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing [s@ivr-2:9] Set("SIP/1747xxxxxxx-09bacf30", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing [s@ivr-2:10] BackGround("SIP/1747xxxxxxx-09bacf30", "custom/Welcome") in new stack
    -- Executing [s@ivr-2:11] WaitExten("SIP/1747xxxxxxx-09bacf30", "|") in new stack

Running the test call (7777) gives me this:

  -- Executing [7777@from-internal:1] Goto("SIP/101-09b7a250", "from-pstn|s|1") in new stack
    -- Goto (from-pstn,s,1)
    -- Executing [s@from-pstn:1] Set("SIP/101-09b7a250", "__FROM_DID=s") in new stack
    -- Executing [s@from-pstn:2] GotoIf("SIP/101-09b7a250", "1 ?cidok") in new stack
    -- Goto (from-pstn,s,4)
    -- Executing [s@from-pstn:4] NoOp("SIP/101-09b7a250", "CallerID is "device" <101>") in new stack
    -- Executing [s@from-pstn:5] Goto("SIP/101-09b7a250", "ivr-2|s|1") in new stack
    -- Goto (ivr-2,s,1)
    -- Executing [s@ivr-2:1] Set("SIP/101-09b7a250", "LOOPCOUNT=0") in new stack
    -- Executing [s@ivr-2:2] Set("SIP/101-09b7a250", "__DIR-CONTEXT=default") in new stack
    -- Executing [s@ivr-2:3] Set("SIP/101-09b7a250", "_IVR_CONTEXT_ivr-2=") in new stack
    -- Executing [s@ivr-2:4] Set("SIP/101-09b7a250", "_IVR_CONTEXT=ivr-2") in new stack
    -- Executing [s@ivr-2:5] GotoIf("SIP/101-09b7a250", "0?begin") in new stack
    -- Executing [s@ivr-2:6] Answer("SIP/101-09b7a250", "") in new stack
    -- Executing [s@ivr-2:7] Wait("SIP/101-09b7a250", "1") in new stack
    -- Executing [s@ivr-2:8] Set("SIP/101-09b7a250", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing [s@ivr-2:9] Set("SIP/101-09b7a250", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing [s@ivr-2:10] BackGround("SIP/101-09b7a250", "custom/Welcome") in new stack
    -- <SIP/101-09b7a250> Playing 'custom/Welcome' (language 'en')

I apologize if I’ve missed / overlooked anything. I’m still learning when it comes to asterisk. Thanks for any help!

-Carl Blakemore

2

Somewhere along the way I had put in my allow statement for my trunk “ilbc”. I’ve come to learn that asterisk doesn’t have a license for ilbc, but X-Lite does. So when they were dumping it straight out to the extension, it worked fine, but when asterisk had to deal with it, since it didn’t have a license for ilbc, it couldn’t handle it. Removing ‘ilbc’ from the list for ‘allow’ fixed the problem.